search for: callins

hi,Jimmy Godbout when +86 13666666666 make a call to -> +86 020 87654321xxxx -> asterisk the CALLERID(num) will show the caller number +8613666666666 the ${EXTEN} is the dialed number 87654321 i will try the CALLERID(dnid) tomorrow, will this get the whole dialed number 87654321xxxx ? i don't know whether the sp will send me the whole number.... ------------------ ssmax

...ing function. The call is to be inserted at a random location in every function in the code. The call does get inserted, however, the module crashes later (after doFinalization()). The crash happens only sometimes, not always. These are the kind of instructions that I see in the generated code. %callins = call void @myfunc(), !dbg !285 %callins = call void @myfunc() The stack trace shows the crash has something to do with debug information. Do I need to add metadata or debug info to the newly added call instruction? Any pointers on how I can do that? Here is the code snippet and the stack trac...

Hello, I lookin' for a call in number from UK or USA. Can somebody offers me a peering for this or specify any sip provider that offers this thing? Thank you very much, Jonson.

Hi when I call the function below in R, i get the error: Object 'pairlist' can't be converted to 'double'. #include <R.h> #include <Rdefines.h> #include <Rmath.h> SEXP CSimPoisson(SEXP lambda, SEXP tgrid, SEXP T2M, SEXP Ni, SEXP NT) { double sign, EVar; double *xlambda, *xtgrid, *xT2M, *xNi, *xNT, *xtau; SEXP tau; int ltgrid =0; int i = 0; int j = 0;

Hi all i have just set up a asterisk in china, using DE410P and one E1 line and get a phone number like: +86 020 87654321 from my sp when somebody dial +86 020 87654321 , the asterisk will get the call in number by ${EXTEN} variable, but it can only get 87654321, no area code . when someone dial +86 020 87654321xxxx , xxxx means 4 digits, the phone can call in, and the

I have a zyxel wifi phone and it works just great registering and callin on * but when it comes to menus it just not working (buttons dont change menus) . Anybody else have this problem? Need help. thanks. -------------- next part -------------- A non-text attachment was scrubbed... Name: Jim Van Eeckhoutte.vcf Type: text/x-vcard Size: 136 bytes Desc: not available Url :

Asymmetric wrote: [SNIP] > > I don't know (don't use ices) but I thought I should inform you.. taking > user requests may cause you to violate the RIAA rules regarding that > topic, depending on how you handle it. Be careful. ;) > It's for my office... I'd love to play my CDs on my radio, but it would disturb those that don't like my music. Some do, so

Hello List. I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for "?VoIP residential services"? if yes, how? if no,

Hi all I get a Cisco 1841 router, connecting the office through a CHAP PPPoE link: [...] interface Dialer1 ip address negotiated ip mtu 1492 ip nat outside ip virtual-reassembly encapsulation ppp ip tcp adjust-mss 1452 dialer pool 1 dialer-group 1 no cdp enable ppp authentication chap callin ppp chap hostname username at isp ppp chap password 0 s0meStr0ngP4sswd [...] As

> If I do this publicly, I've already researched the royalty situation, > etc. It's not all that complicated in that regard... radio stations > already allow callin shows. This is just a variation. Online radio is COMPLETELY DIFFERENT than on-the-air radio. You MUST follow the RIAA guidelines for non-interactive broadcasts for online radio, and you do not have to follow

Hi, i have one Asterisk with a Digium E1 card, and a Meridian Nortel Release 11. I need to connect both of them. We are using MFC/R2 for this.. The Diagram: [ NORTEL ] ( AMI ) ------------------------------------------------ (DIGIUM) [ ASTERISK] we have green light at the digium card, and at asterisk we see all 31 channels as idle. But when i want to recive a call from the Nortel to the

Hi, i have one Asterisk with a Digium E1 card, and a Meridian Nortel Release 11. I need to connect both of them. We are using MFC/R2 for this.. The Diagram: [ NORTEL ] ( AMI ) ------------------------------------------------ (DIGIUM) [ ASTERISK] we have green light at the digium card, and at asterisk we see all 31 channels as idle. But when i want to recive a call from the Nortel to the

Hello everyone, Just curious if ices supports dynamic playlists... I would like to generate one on the fly based on user requests etc. I'm startnig to go through icedj's source code to see how they use shout. I'd just use that, but I'm having trouble with shout. Thanks, Sujal -- ---- Sujal Shah --- sujal@sujal.net --- http://www.sujal.net Now Playing:

Jack Moffitt wrote: >>If I do this publicly, I've already researched the royalty situation, >>etc. It's not all that complicated in that regard... radio stations >>already allow callin shows. This is just a variation. >> > Wow, this must have just happened in the last 12 months (I last looked into this about a year ago). Can you point out a URL to the

Am 07.01.2016 um 10:55 schrieb Frank: > On Wed, 2016-01-06 at 17:03 +0100, Juergen Sauer wrote: Thx, 4answer. :) >> with in my sip.conf, I have got for this hardphone: >> [...] >> [hard1] >> username=hard1 >> secret=correct-and-three-times-checked-4-digit-pin > > In most cases, there is no need to set the "username=" option. The

Its been a few days, I was wondering if anyone else has any ideas on how to get this to work? If not, could I ask a the direct question as to if anyone here has (or knows someone who has) successfully got something like a Cisco Router to successfully establish an ISDN data/Internet connection through the Asterisk PBX?

Have you got termination set correctly? I have a B410P working with 2 x NT and 2 x TE ports successfully. I had to turn the 100ohm termination on on the NT ports (even though I have them set as PTP in mISDN.conf). HTH -->> -----Original Message----- -->> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- -->> bounces at lists.digium.com] On

Is there a little setup display right on the box? Just asking because I have seen that on some boxes. Cheers, Boris. On Sun, Sep 18, 2016 at 12:15 AM, Keith Keller < kkeller at wombat.san-francisco.ca.us> wrote: > On 2016-09-17, Alice Wonder <alice at domblogger.net> wrote: > > > > Okay if it requires DHCP this might be out, I'm currently out of town > >

Good Evening, The company I work for is attempting to connect an Cisco ISDN Router to an OpenVOX B200P BRI Card so that we can get it to dial out across an existing ISDN PRI Line also installed in the Asterisk PBX Everything has compiled successfully and the BRI card has been detected when mISDN is started: # misdnportinfo Port 1: NT-mode BRI S/T interface port (for phones) -> Interface

I recently bought a new UPS, and I''m attempting to use nut to monitor it. Following setup instructions everything seemed to go well until it came to testing the connection, which failed. There is just one serial connector on the computer, so I set it to monitor /dev/ttyS0. Either that is wrong, or communication is failing. I''ve been told to try minicom to monitor it,